[Asterisk-Users] codec negotiation
Eduardo Goncalves
eduardo at acenet.com.br
Tue Dec 16 11:08:00 MST 2003
Hi list,
I'm with a little problem on codec negotiation between a cisco827 and
asterisk.
My sip.conf is like that:
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
amaflags = default
allow=g729
allow=gsm
allow=alaw
allow=ulaw
;disallow=all
and cisco like that:
dial-peer voice 6 voip
destination-pattern 0T
session protocol sipv2
session target ipv4:<asterisk-ip>
dtmf-relay rtp-nte
codec g711alaw
no vad
!
When I try to make a call, cisco shows codec g711alaw, but asterisk
shows codec g729A (i have the licenses) and there is no audio. When I
try disallow=g729, the same occurs, but this time asterisk shows codec
gsm.
The only way to make a call is allowing only alaw. But this is not
convenience, since i need to use g279 with another endpoint (working
ok).
Why this negotiation problem happens?
Thanks
Eduardo
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